| Index: webrtc/voice_engine/file_player.h
|
| diff --git a/webrtc/voice_engine/file_player.h b/webrtc/voice_engine/file_player.h
|
| deleted file mode 100644
|
| index cb14c8f7523930d1eef90b0305e07bea9ecceebe..0000000000000000000000000000000000000000
|
| --- a/webrtc/voice_engine/file_player.h
|
| +++ /dev/null
|
| @@ -1,80 +0,0 @@
|
| -/*
|
| - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
| - *
|
| - * Use of this source code is governed by a BSD-style license
|
| - * that can be found in the LICENSE file in the root of the source
|
| - * tree. An additional intellectual property rights grant can be found
|
| - * in the file PATENTS. All contributing project authors may
|
| - * be found in the AUTHORS file in the root of the source tree.
|
| - */
|
| -
|
| -#ifndef WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
|
| -#define WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
|
| -
|
| -#include <memory>
|
| -
|
| -#include "webrtc/common_types.h"
|
| -#include "webrtc/modules/include/module_common_types.h"
|
| -#include "webrtc/typedefs.h"
|
| -
|
| -namespace webrtc {
|
| -
|
| -class FileCallback;
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| -
|
| -class FilePlayer {
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| - public:
|
| - // The largest decoded frame size in samples (60ms with 48kHz sample rate).
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| - enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 48 };
|
| - enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 };
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| -
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| - // Note: will return NULL for unsupported formats.
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| - static std::unique_ptr<FilePlayer> CreateFilePlayer(
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| - const uint32_t instanceID,
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| - const FileFormats fileFormat);
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| -
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| - virtual ~FilePlayer() = default;
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| -
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| - // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
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| - // will be set to the number of samples read (not the number of samples per
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| - // channel).
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| - virtual int Get10msAudioFromFile(int16_t* outBuffer,
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| - size_t* lengthInSamples,
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| - int frequencyInHz) = 0;
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| -
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| - // Register callback for receiving file playing notifications.
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| - virtual int32_t RegisterModuleFileCallback(FileCallback* callback) = 0;
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| -
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| - // API for playing audio from fileName to channel.
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| - // Note: codecInst is used for pre-encoded files.
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| - virtual int32_t StartPlayingFile(const char* fileName,
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| - bool loop,
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| - uint32_t startPosition,
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| - float volumeScaling,
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| - uint32_t notification,
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| - uint32_t stopPosition,
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| - const CodecInst* codecInst) = 0;
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| -
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| - // Note: codecInst is used for pre-encoded files.
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| - virtual int32_t StartPlayingFile(InStream* sourceStream,
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| - uint32_t startPosition,
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| - float volumeScaling,
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| - uint32_t notification,
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| - uint32_t stopPosition,
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| - const CodecInst* codecInst) = 0;
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| -
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| - virtual int32_t StopPlayingFile() = 0;
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| -
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| - virtual bool IsPlayingFile() const = 0;
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| -
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| - virtual int32_t GetPlayoutPosition(uint32_t* durationMs) = 0;
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| -
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| - // Set audioCodec to the currently used audio codec.
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| - virtual int32_t AudioCodec(CodecInst* audioCodec) const = 0;
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| -
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| - virtual int32_t Frequency() const = 0;
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| -
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| - // Note: scaleFactor is in the range [0.0 - 2.0]
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| - virtual int32_t SetAudioScaling(float scaleFactor) = 0;
|
| -};
|
| -} // namespace webrtc
|
| -#endif // WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
|
|
|